Josh,
Thank you for starting to work on this. It's something very important for me. I
build platforms where many Asterisk servers handle the same call and we're
hunting all latency we can.
I have no intelligent comments on your work, more than what you write seems
correct - we should only do
http://forums.asterisk.org/viewtopic.php?f=1t=88967
http://stackoverflow.com/questions/20966934/any-help-every-time-ast-frame-dtmf-end-
asterisk-11-5-1-app-confbridge-c
In Asterisk11.5.1/funcs/func_frame_trace.c it works fine,
but in my code issue : frame-frametype
. BEGIN and END are not
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Please also check the sender IP address in the IP packet.
Hello,
I have a question regarding sRTP with Asterisk. I posted this question on the
Asterisk wiki and was suggested to post it on the Developer mailing list. I am
working on an encryption project. The project consists of using my own
encryption method to encrypt SIP phone calls using an
On 07 Jan 2014, at 15:12, Dylan Herman herma...@pascack.k12.nj.us wrote:
Hello,
I have a question regarding sRTP with Asterisk. I posted this question on the
Asterisk wiki and was suggested to post it on the Developer mailing list. I
am working on an encryption project. The project
On 07 Jan 2014, at 15:55, SVN commits to the Digium repositories
svn-comm...@lists.digium.com wrote:
if (endpoint-nat.rewrite_contact (contact =
pjsip_msg_find_hdr(rdata-msg_info.msg, PJSIP_H_CONTACT, NULL))
- (PJSIP_URI_SCHEME_IS_SIP(contact-uri) ||
Olle E. Johansson wrote:
On 07 Jan 2014, at 15:55, SVN commits to the Digium
repositoriessvn-comm...@lists.digium.com wrote:
if (endpoint-nat.rewrite_contact (contact =
pjsip_msg_find_hdr(rdata-msg_info.msg, PJSIP_H_CONTACT, NULL)) -
(PJSIP_URI_SCHEME_IS_SIP(contact-uri) ||
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Ship it!
I think this unconditional behavior is fine since
On Tue, Jan 7, 2014 at 9:27 AM, Joshua Colp jc...@digium.com wrote:
Olle E. Johansson wrote:
On 07 Jan 2014, at 15:55, SVN commits to the Digium
repositoriessvn-comm...@lists.digium.com wrote:
if (endpoint-nat.rewrite_contact (contact =
pjsip_msg_find_hdr(rdata-msg_info.msg,
On Tue, Jan 7, 2014 at 5:03 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Jan 7, 2014 at 9:27 AM, Joshua Colp jc...@digium.com wrote:
Olle E. Johansson wrote:
On 07 Jan 2014, at 15:55, SVN commits to the Digium
repositoriessvn-comm...@lists.digium.com wrote:
if
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Review request for Asterisk Developers.
Bugs: ASTERISK-22834 and
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Review request for Asterisk Developers.
Bugs: AST-1258
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Review request for Asterisk Developers.
Bugs: ASTERISK-22946
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Review request for Asterisk Developers.
Repository: Asterisk
On 14-01-07 10:10 AM, Olle E. Johansson wrote:
On 07 Jan 2014, at 15:55, SVN commits to the Digium repositories
svn-comm...@lists.digium.com wrote:
if (endpoint-nat.rewrite_contact (contact = pjsip_msg_find_hdr(rdata-msg_info.msg,
PJSIP_H_CONTACT, NULL))
-
Paul Belanger wrote:
IMO, these are also perfect candidates for parsing unit tests. It would
have been great to also see us add something to better increase out code
coverage.
PJSIP has unit tests already which covers the URI parser. I just did a
rough count and there's 39 different ones.
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Ship it!
Ship It!
- wdoekes
On Jan. 7, 2014, 7:28 p.m.,
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The patch is mostly good. One thing that should be done though
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Ship it!
Your findings regarding the to-tag with chan_sip.c
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I'm curious, has the reporter of ASTERISK-22946 confirmed that
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Ship it!
Ship It!
- Mark Michelson
On Jan. 7, 2014, 6:33
Thank you for responding.
I already have the libsrtp on my system, I believe, since I have the same files
that were on the libsrtp github. Are you saying that I could just write my code
in one of the scripts with the encryption algorithms leaving it under its
original name of the algorithm?
8 jan 2014 kl. 01:24 skrev Dylan Herman herma...@pascack.k12.nj.us:
Thank you for responding.
I already have the libsrtp on my system, I believe, since I have the same
files that were on the libsrtp github. Are you saying that I could just write
my code in one of the scripts with the
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